Re: [asterisk-users] What is SIP Early Media useful for ?
I've always wondered if early media with video would allow to see who is calling before answering. Is it possible? Is anybody doing this? On Wed, Feb 3, 2016 at 11:56 PM, Olivier wrote: > Hello, > > Could you help me to summarize what is SIP Early Media useful for ? > > I was thinking of: > - Passing error messages to caller, > - Custom ringing tones to caller. > > Did I miss something ? > > Best regards > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Receiving Messages and Extensions Config for WebRTC
Hi All,
I have configured WebRTC according to the install document.
The clients register correctly. I'm use SIPjs.
The clients are able to send messages to the server. The SIP debug shows
the messages being received.
However I'm stumped for directions on how to route the messages between the
clients.
Asterisk 11.11.0
Here is my client sip config:
[1060]
type=friend
username=1060 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=fee50 ; The SIP Password for SIP.js
;encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to use ICE for this peer
context=default ; Tell Asterisk which context to use when this peer is
dialing
directmedia=no ; Asterisk will relay media for this peer
transport=ws ; Asterisk will allow this peer to register on UDP or
WebSockets
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
nat=force_rport,comedia
accept_outofcall_message=yes
outofcall_message_context=messages
;dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
;dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
;dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your
DTLS cert file is
;dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your
DTLS private key is
;dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when
setting up DTLS
[1061] ; This will be the legacy SIP client
type=friend
username=1061
host=dynamic
secret=fee50
;encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to use ICE for this peer
context=default ; Tell Asterisk which context to use when this peer is
dialing
directmedia=no ; Asterisk will relay media for this peer
transport=ws ; Asterisk will allow this peer to register on UDP or
WebSockets
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
nat=force_rport,comedia
accept_outofcall_message=yes
outofcall_message_context=messages
;dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
;dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
;dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your
DTLS cert file is
;dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your
DTLS private key is
;dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when
setting up DTLS
Here is my extensions config: (I guess this is the wrong way to go, but any
pointers are appreciated).
[messages]
exten => 1060,1,Dial(SIP/1060) ; Dialing 1060 will call the SIP client
registered to 1060
;exten => 1061,1,Dial(SIP/1061) ; Dialing 1061 will call the SIP client
registered to 1061
exten => 1061,1,NoOp(Message from: ${MESSAGE(from)})
same => n,NoOp(Message to: ${MESSAGE(to)})
same => n,NoOp(Message body: ${MESSAGE(body)})
same => n,MessageSend(sip:1061@254.248.223.23:$[SIPPEER(1061,port)])
same => n,NoOp(Message send status: ${MESSAGE_SEND_STATUS})
same => n,Hangup()
Thank you
Vivian
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Re: [asterisk-users] Looking for Asterisk Consultants & Experts
Hi there, I can help you, please explain what do you need as detailed as possible for an estimate. Best regards, Alan On Wed, Sep 2, 2015 at 8:40 PM, Shahid H wrote: > Hello, > > Can someone recommend me where is best place to find Asterisk > Expert/Consultant for freelance work? > > If you are interested to work as a freelancer, you can email me directly. > > Thanks > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Video through IAX2 trunks
I have a couple of Asterisk 13.4 servers with an IAX2 trunk between them: [phone1] <--pjsip--> [server1] <--iax2 trunk--> [server2] <--pjsip--> [phone2] With this setup, audio calls work fine, but video doesn't work: I get a black window and if I remember correctly, I was getting white noise in one direction (not sure if the noise thing is asterisk's fault or the phone's). I do see the video packets going through when I enable debugging on iax2, so this is strange. I tested with linphone, pjsua, and a Grandstream gxv3240. Video calls that don't go through the iax2 trunk work fine (eg: both phones in the same server). If I change the trunk to use pjsip, video works fine, but then I have other problems that am still working on. I'd rather keep using iax2 for my trunks for a while, so my question is: Is anybody using video in this kind of setup? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any video applications available
hI Check WebRTC http://www.youtube.com/watch?v=E8C8ouiXHHk 2013/2/28 Jimmy Chang(Gmail) > We found this URL: > http://sourceforge.net/**projects/asteriskvideo/<http://sourceforge.net/projects/asteriskvideo/> > But these applications seem too old for Asterisk 11. > > Are there any video applications for Asterisk 11? > We need these applications to implement IVVR. > > Or any other solution is to be appreciated. > > Thanks in advance. > > -- > __**__**_ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users> > -- *Alan Sánchez Torres* *Telefono:* 940484090 *Email: *asanc...@vtelcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for "solid state" like PC suitable for Asterisk
On 10/05/12 09:49, Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? http://www.rowetel.com/blog/?page_id=445 HTH Al -- Libertus Solutions http://www.libertus.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?
On 26/08/11 19:02, linux guy wrote: I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home asterisk system. We've been using the Siemens Gigaset 685IP range for over three years and I'm (still) very pleased with them: http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/ -- The Open Learning Centre http://www.theopenlearningcentre.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
On 26/08/11 12:28, linux guy wrote: Great discussion, all of it. Thanks, people. How much power does the home asterisk box need ? Not much :-) I've been running our phone system and home media/storage network on a VIA C7 cpu based home build that I *downclocked* to 1Ghz from 1.2Ghz for about three years now. Al -- The Open Learning Centre http://www.theopenlearningcentre.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail hangs up
Hangup("SIP/203-0003", "") in new
stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'SIP/203-0003' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/203-0003'
--- END ---
Here is my sip_additional.conf entry for the extension I used above (302):
--- START ---
[203]
deny=0.0.0.0/0.0.0.0
secret= (blanked for security reasons)
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/203
mailbox=203@default
permit=0.0.0.0/0.0.0.0
callerid=device <203>
callcounter=yes
faxdetect=no
--- END ---
Here is the entry for extension 203 in extensions_additional.conf:
--- START ---
exten => 203,1,Macro(exten-vm,203,203)
exten => 203,n,Goto(vmret,1)
exten => 203,hint,SIP/203
exten => ${VM_PREFIX}203,1,Macro(vm,203,DIRECTDIAL,${IVR_RETVM})
exten => ${VM_PREFIX}203,n,Goto(vmret,1)
exten => vmb203,1,Macro(vm,203,BUSY,${IVR_RETVM})
exten => vmb203,n,Goto(vmret,1)
exten => vmu203,1,Macro(vm,203,NOANSWER,${IVR_RETVM})
exten => vmu203,n,Goto(vmret,1)
exten => vms203,1,Macro(vm,203,NOMESSAGE,${IVR_RETVM})
exten => vms203,n,Goto(vmret,1)
--- END ---
and here is the entry for extension 203 in voicemail.conf:
--- START ---
203 => 12345,Test,,,attach=no|saycid=no|envelope=no|delete=no
--- END ---
Hopefully the above helps? Please advise if there is any further info you
might need, or what you would like me to try.
Thanks! :-)
-Alan
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Re: [asterisk-users] asterisk + openBTS
On 19/08/10 18:20, equis software wrote: > I want to know about asterisk and openBTS > If anybody made any test and experience... This island runs it's GSM network on OpenBTS: http://www.niueisland.com/ This was the place he presented about. Read the blog here: http://openbts.sourceforge.net/NiuePilot/ HTH Al -- The Open Learning Centre http://www.theopenlearningcentre.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk + openBTS
On 19/08/10 18:20, equis software wrote: > I want to know about asterisk and openBTS > If anybody made any test and experience... I saw a presentation a few months ago where one of the openBTS project founders talked about one early system they set up on a very small and remote Pacific island along with Asterisk. If I can remember/find anything more I'll post here. Cheers Alan -- The Open Learning Centre http://www.theopenlearningcentre.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] install asterisk
On 13/08/10 14:08, Albert Bonomo wrote: > This time, the server was up with Fedora 13. No problem. > Well not so fast. Yes problem !!! I can not install it !!! Fedora is *not* a server operating system and not one I would choose to run asterisk on. I would recommend using either CentOS or a Debian/Ubuntu Server build without X11 and all the other cruft that comes with a Desktop OS - which is what Fedora is. I'd also consider how the packagers of CentOS or Debian chose to install Asterisk before deciding on using a pre-packaged implementation. CMMI and your own understanding of where/why and how it is installed will probably help you more in the longer run. HTH Al -- The Open Learning Centre http://www.theopenlearningcentre.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FYI: Seen the 2600Hz announcement?
http://gigaom.com/2010/08/03/2600hz-project/ -- The Open Learning Centre http://www.theopenlearningcentre.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX softphone
On 02/08/10 17:35, Ronaldo Zacarias Afonso wrote: > Hi all, > > Can some one suggest me an IAX client for Linux and Windows? > I used KIAX once, but know it seems complicated to have it working on Ubuntu. This one is great on Ubuntu/Linux. http://www.sflphone.org/ Unfortunately I know not about Windows though, I never use it. Cheers Al -- The Open Learning Centre http://www.theopenlearningcentre.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 69, Issue 16
Hello All: I saw there are app_fax and app_chanspy modules in 1.6.2.6, but there is NO sample configure file for them. Is anybody know how to use them, or where is the documentation for them? Thanks -- Refer to: http://www.microsuncn.com Best Regards Alan Zheng -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Asterisk 1.6 and DECT Phones
On 23/02/10 08:38, Randy R wrote: > On Tue, Feb 23, 2010 at 9:23 AM, Alan Lord (News) > wrote: >> Another vote for the Siemens Gigaset range. Been using the S685IP almost >> since the day it was released here in the UK. Nice handsets, great voice >> quality, but as others have said the UI can be a bit slow. > > Alan, don't forget the link to the discussion on your excellent site: > > http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/ Thanks for the plug Randy :-) Al -- The Open Learning Centre http://www.theopenlearningcentre.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Asterisk 1.6 and DECT Phones
On 22/02/10 16:18, --[ UxBoD ]-- wrote: > Hi, > > looking for your valued input on suitable suggestions for high quality VoIP > DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and > looking to a new manufacturer. > Another vote for the Siemens Gigaset range. Been using the S685IP almost since the day it was released here in the UK. Nice handsets, great voice quality, but as others have said the UI can be a bit slow. Do watch out for the last firmware release though. Siemens had some trouble with that although I personally haven't experienced the issues reported: http://www.mgraves.org/voip/2009/11/gigaset-firmware-update-released/ It looks like a new firmware release is imminent: http://www.mgraves.org/voip/2010/02/gigaset-news-new-beta-firmware-release/ HTH Alan -- The Open Learning Centre http://www.theopenlearningcentre.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Siemens Gigaset + Asterisk Query?
When you configure the Siemens gigaset handsets (I have S685IP), there is a single option for all handsets to use either the POTS interface or VOIP as the default outbound destination - you then need to add a dial suffix if you want to use an alternate outbound route. Does anyone have any suggestions as to how to make just *one* of the DECT handsets only use the POTS but others default to their Asterisk SIP subscriptions? The POTS is on the Gigaset base station & not on the Asterisk server. TIA. Al -- The Open Learning Centre http://www.theopenlearningcentre.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for an Asterisk supervision (status viewer) tool
On 11/11/09 10:00, Steve Howes wrote: > > On 11 Nov 2009, at 09:06, Alan Lord (News) wrote: >> >> "Warning: Your browser may not be able to handle this site! Please >> upgrade your browser to the latest version of Internet Explorer, >> Firefox, Mozilla or Netscape. >> >> Sorry for any inconvenience. >> >> - The OrderlyQ Team." >> >> I am using Firefox 3.5.4 on Ubuntu. It is up-to-date. This kind of >> thing >> just makes me leave your site without investigating further. > > Website is probably too retarded to recognise the non-trademark names > that various organisations use for Firefox. Ubuntu have a deal with Mozilla and do distribute "Firefox". It isn't Iceweasel or whatever... Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for an Asterisk supervision (status viewer) tool
On 10/11/09 18:19, Christina Casey wrote: > Hi Klaus, > > Yes all the below is possible/easy with the OrderlyStats call centre > management and reporting tool. > > It's a free download - please see http://www.orderlyq.com/orderlystats.html > > Kind regards, > > Christina Casey > Accounts Manager > Orderly Software Ltd. When I visit your website I get a very annoying alert claiming: "Warning: Your browser may not be able to handle this site! Please upgrade your browser to the latest version of Internet Explorer, Firefox, Mozilla or Netscape. Sorry for any inconvenience. - The OrderlyQ Team." I am using Firefox 3.5.4 on Ubuntu. It is up-to-date. This kind of thing just makes me leave your site without investigating further. Also, what does the alert mean by Mozilla or Netscape? Mozilla isn't a browser, and Netscape is basically dead. From the Wikipedia: "Tom Drapeau, director of AOL's Netscape Brand, announced that the company would stop supporting Netscape software products as of March 1, 2008.[2] The decision met mixed reactions from communities, with many arguing that the termination of product support is significantly belated. Internet security site Security Watch stated that a trend of infrequent security updates for AOL's Netscape cause the browser to become a "security liability", specifically the 2005-2007 versions, Netscape Browser 8." http://en.wikipedia.org/wiki/Netscape I kindly suggest you get your website fixed. Cheers Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI for asterix management
On 24/10/09 11:05, Steve Howes wrote:
>
> On 24 Oct 2009, at 10:52, giancarlo lombardo wrote:
>> at the moment machine is standalone, so i need to start GUI from
>> console.
>
> There is no Asterisk GUI that runs like that. You could install X and
> Firefox but thats just a bit retarded. Plug in a network cable! Its
> not like its not going to need one eventually.
If it's a true server with no X you could just use a text browser like
Lynx or Links. Then browse to http://localhost{:port}/gui url
But to be frank, the Op will learn more by editing the configuration
files directly.
HTH
Alan
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Re: [asterisk-users] OT - DECT SIP Phones
On 23/10/09 08:34, --[ UxBoD ]-- wrote: > > S685 set turned up the other day and have had a good chance to try it out ... > Voice quality is definitely superior to the M3 though I guess that will be > addressed in the M9 with G722 support. > > Impressions of the Siemens phone :- > > Pros > > Great Voice Quality > Easy to configure > Good range > Ability to transfer address books > Good build quality > > Cons > > Keys feel fiddly > Keys response time when dialling is very slow > When call connection is made a audible beep is heard which cuts of the first > .5 second of a VM message > DTMF does not all transfer correctly > > On the whole it seems a very capable phone and very well laid out. For my > personal needs it just does not feel right with how the keys react. I agree with the response time on the keys, it does seem to be quite slow but I have got used to that now - after 1 1/2 years :-) Will the M9 also have an analogue interface? That is one of the main reasons for my choosing this phone. It's a great dual home/home-office phone because of that. Cheers Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - DECT SIP Phones
On 17/10/09 15:02, --[ UxBoD ]-- wrote: > Hi, > > I have three Snom M3s at the moment but getting pretty fed up with the issues > :( I am UK based and would be interested to hear of other peoples > recommendations. Key features :- > > * VM Notification > * Good Range > * G729 codec support > * Common/Private Address Books per Handset(s) Siemens Gigaset: http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/ One of the most popular posts on my blog over the last 1 1/2 years. It still gets lots of hits from people looking for info on them. FYI We have two sets in our network - they haven't missed a beat since installation. HTH Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] x100p card
Hi All: I have a x100p card and asterisk installed to my computer, can I use this card and a softphone to call other phone? Thanks -- Refer to: http://www.microsuncn.com Best Regards Alan Zheng ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk + Skype deployment
Just FYI Really, nothing to do with me... http://www.thevarguy.com/2009/10/01/systems-integrator-dials-skype-for-asterisk/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choose IAX or SIP trunking?
On 01/10/09 00:57, Kirill 'Big K' Katsnelson wrote: > Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID > calls, originating and transferring. > > A provider offers both SIP and IAX trunking. Cateris paribus, what is > the preferred solution to choose? What points to consider? We use IAX trunks from our provider primarily as they are so much easier to configure and you only need one port open on your firewall/nat gateway. SIP needs hundreds, if not thousands of open ports IIUC. HTH Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP solutions
On 26/09/09 19:42, Hans Witvliet wrote: > What you can do (perhaps not the best solution...) is having one > asterisk server behind your firewall, serving all your local > sip-clients. And another at the other side of the firewall, only for > serving remote clients. And have both systems talking to each other with > IAX instead of SIP > > In that case you _only_ have to allow port 4569 for IAX instead of 5060 > and 1...2 for SIP Hmmm, has anyone tried SIP over a VPN? We are thinking of testing this but haven't yet... Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel kernel configuration error on vmware
On 12/09/09 08:03, DHAVAL INDRODIYA wrote: > hello > while i try to compile zaptel > > it gives following error to me > > you do not appear to have the sources for the 2.6.27-7-server kernel > installed You will need to install the source tree (or at least the headers) for the running kernel. On debian there is a package called kernel-headers (usually with the version number). I don't use Centos/Red hat much but `yum install kernel-headers` might be worth a try or use Google :-) Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - DECT handset with Line key
On 18/08/09 08:08, Olivier wrote: > Hi, > > I need to replace digital handsets in offices where there cabling is > appareantly not Ethernet-compliant. > Today's usage is to press a key to toggle between private ou public line > before issuing an outgoing call. > > Are you aware of a DECT handset (to overcome cabling limitations) that > mimic this line-key behaviour ? > For instance, acceptable behaviours would be to dial number string and > press Green key to send a call using private line or Blue key to send it > using public line. > > Alternative is to prepend dial string with a specific code for private > calls (for example, *3100123456789 meaning call 00123456789 using > private line). We use Siemens S685IP DECT handsets. I think they can do what you need. Here's a review of them I did last April. http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/ You don't say where you are. I do not believe these h/s are available in North America. Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server linux requirements
On 04/08/09 23:57, Miguel Molina wrote: > Edwin Quijada escribió: >> It depends about your traffic. >> But myabe and I guess Core 2Duo 4Gb ram , Sata 160Gb >> +_ >> >> > It's pretty well known that asterisk is CPU intensive, not RAM > intensive. It think 4GB is much more than enough. BTW, if your asterisk > is consuming more than 150-200MB of RAM and growing... suspect of memory > leaks. I once read on this list than seeing 500MB of memory usage for an > asterisk instance is *huge*. Huh? CPU load is generally down to things like transcoding... If you are switching just G.711 (a/ulaw) then CPU load is negligible in my experience. We've seen Asterisk systems for tens of users that run on tiny embedded processors or the low power VIA C7 type CPUs. I think it is more important to determine what level of reliability you need and engineer your server(s) around that. If you want "Five 9s" you'll need to buy the right hardware and probably duplicate it. Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TLS Manager
On 25/07/09 00:08, John A. Sullivan III wrote: > Hello, all. After many pages of googling and testing in the lab, I'm > still a bit perplexed about how to implement tls protection for the > asterisk manager. manager.conf allows one to specify the cert file but > one normally must also specify the private key file. If I simply enter > the cert file: > > sslenable=yes > sslbindport=5038 > sslbindaddr=172.x.x.8 > sslcert=/etc/pki/tls/certs/pbxc.pem ; path to the certificate. > ; sslcipher= > > It errors as I expect it would: > > pbx*CLI> manager reload >== Parsing '/etc/asterisk/manager.conf': == Found > SSL cert error > > How does one specify the private key for the manager.conf file? Thanks - > John Not quite the same thing I know, but it might help. I use stunnel for the AMI so the connection is transported in a SHH tunnel. It's quite easy to setup. Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I create an IVR/Dial Group that worksproperly?
On 19/07/09 04:50, Maxi Belino wrote: > > It get's really hard to to try and deal with all the possibilities > reliably. > > IMHO, the "Destination" field *should* contain simply the number of the > destination ext. of the call; as it rightly does when digits are > actually dialled by the caller. Why it doesn't when the call is > generated by the dialplan IVR is just plain inconsistent. > > Alan > > Hi Alan, > > did you find the way to solve this issue? > > regards, > Maxi Unfortunately no. The more I think about this, the more I think it is a bug. I mailed the asterisk-dev list to see what they think about it, but I haven't seen *any* posts appear on it since Friday so I wonder if it is currently unavailable or something. I will try again later today. Also, if you discover anything do please reply to this thread. Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Truecall
On 18/07/09 00:35, Gavin Henry wrote: > This has to be an Asterisk based appliance no? > > http://www.truecall.co.uk/acatalog/trueCall_Features.html I saw this on the TV the other night. Couldn't believe how the "dragons" all thought it was such a cool idea. I was shouting at the telly saying "You could do that with Asterisk very easily"... Granted, if he's made the box, built it on an embedded SoC device then fair play, but he needs to have something Unique or anyone can do it. Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I create an IVR/Dial Group that worksproperly?
On 17/07/09 17:20, Danny Nicholas wrote:
> Not that this will really help, but in my CDR, I get this find of format
> Xxx incoming_number s context caller_id incoming_tech/line
> target_tech/line function command time1 time2 time3. It seems that
> you could look to the target_tech/line for the information you need.
Yeah I know what you mean. That is the "destination channel" which does
contain something like SIP/101-9u1exdo8, even though the "Destination"
contains just "s".
I am working on some CRM integration code and really don't want to have
to parse this stuff if I can help it. Some of our extensions will/could
be on Zap/ or IAX/context/blah-hsdjgdjf-.
It get's really hard to to try and deal with all the possibilities reliably.
IMHO, the "Destination" field *should* contain simply the number of the
destination ext. of the call; as it rightly does when digits are
actually dialled by the caller. Why it doesn't when the call is
generated by the dialplan IVR is just plain inconsistent.
Alan
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord
> (News)
> Sent: Friday, July 17, 2009 11:10 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] How do I create an IVR/Dial Group that
> worksproperly?
>
> On 17/07/09 16:30, David Backeberg wrote:
>> On Fri, Jul 17, 2009 at 4:22 AM, Alan Lord (News)
> wrote:
>>> * Caller arrives at our main number
>>> * Caller is greeted and then told they can enter an extension number, if
>>> known, or wait and their call will be connected to an available rep.
>>> * The IVR then dials a group of extensions (if the caller didn't enter
>>> one obviously).
>>> * Someone picks up the call and the connection is established and logged.
>>>
>>> Now, I have all of this working apart from the last piece.
>>>
>>> My IVR rings various extensions and I can pick up the call just fine.
>>> But my problem is that the data asterisk records regarding the call is
>>> wrong.
>>>
>>> It correctly identifies the CallerID, but it always records the
>>> destination as "s". Not the extension of, for example my SIP phone (101).
>>
>> Somewhere earlier, you do the very first answer. At that point, you should
> add a
>> NoOp(${EXTEN})
>> Set(WHATIREALLYWANTEDINSTEAD=${EXTEN}
>>
>> and then keep popping out the
>> ${WHATIREALLYWANTEDINSTEAD}
>> value wherever you wanted the original extension before you started
>> jumping all over the place in your dialplan.
>
> I don't really understand what you are saying here. Sorry :-(
>
> When the call first hits * (over an IAX trunk), it gets put into the IVR
> [tolc_menu} at s,1 and the extension in the IAX context is the incoming
> number. So there isn't an EXTEN at this stage. And I do not know
> WHATIREALLYWANTEDINSTEAD because:
>
> a) the caller has not yet dialled an extension, or
> b) I do not know which of us will answer the call.
>
>> As you maybe guessed by now, EXTEN is the immediate, right now
>> extension, and if you make jumps, it will update as you jump around.
>
> Well, yes I understand that. So WTF does the extension not *jump* to 101
> or 202 (or whatever the destination is) when a real person finally
> answers the call?
>
>> And then if you want the WHATIREALLYWANTEDINSTEAD value into your CDR,
>> see the earlier post this week regarding setting arbitrary values into
>> your CDR.
>
> It can't be this hard surely?
>
> We can't be the only firm in the world that doesn't do DDI and just has
> one incoming number?
>
> As I said, if while the caller is in the IVR they dial 101 it works
> properly. But some will not know our extension numbers so the IVR rings
> several handsets and the first one to pick up gets the call. Why isn't
> that information set as the destination EXTEN?
>
> I am beginning to think this is probably a bug. It has nothing to do
> with Macros. I have tried without.
>
> Alan
>
>>> [tolc_menu] ; Welcome and information to callers
>>> exten => s,1,Answer()
>>> exten => s,n,Wait(2)
>>> exten => s,n,Background(welcome-to-tolc) ; Say Hello
>>> exten => s,n,Wait(1)
>>> exten => s,n(tryagain),Background(enter-ext-of-person&or) ; Enter
>>> extension number if known, or
>>> exten => s,n,Background(pls-stay-on-line) ; Trying to connect...
>>> exten => s,n,WaitExten(5)
>>> exten => s,n,Macro(belllord,${AL
Re: [asterisk-users] How do I create an IVR/Dial Group that works properly?
On 17/07/09 16:30, David Backeberg wrote:
> On Fri, Jul 17, 2009 at 4:22 AM, Alan Lord (News)
> wrote:
>> * Caller arrives at our main number
>> * Caller is greeted and then told they can enter an extension number, if
>> known, or wait and their call will be connected to an available rep.
>> * The IVR then dials a group of extensions (if the caller didn't enter
>> one obviously).
>> * Someone picks up the call and the connection is established and logged.
>>
>> Now, I have all of this working apart from the last piece.
>>
>> My IVR rings various extensions and I can pick up the call just fine.
>> But my problem is that the data asterisk records regarding the call is
>> wrong.
>>
>> It correctly identifies the CallerID, but it always records the
>> destination as "s". Not the extension of, for example my SIP phone (101).
>
> Somewhere earlier, you do the very first answer. At that point, you should
> add a
> NoOp(${EXTEN})
> Set(WHATIREALLYWANTEDINSTEAD=${EXTEN}
>
> and then keep popping out the
> ${WHATIREALLYWANTEDINSTEAD}
> value wherever you wanted the original extension before you started
> jumping all over the place in your dialplan.
I don't really understand what you are saying here. Sorry :-(
When the call first hits * (over an IAX trunk), it gets put into the IVR
[tolc_menu} at s,1 and the extension in the IAX context is the incoming
number. So there isn't an EXTEN at this stage. And I do not know
WHATIREALLYWANTEDINSTEAD because:
a) the caller has not yet dialled an extension, or
b) I do not know which of us will answer the call.
> As you maybe guessed by now, EXTEN is the immediate, right now
> extension, and if you make jumps, it will update as you jump around.
Well, yes I understand that. So WTF does the extension not *jump* to 101
or 202 (or whatever the destination is) when a real person finally
answers the call?
> And then if you want the WHATIREALLYWANTEDINSTEAD value into your CDR,
> see the earlier post this week regarding setting arbitrary values into
> your CDR.
It can't be this hard surely?
We can't be the only firm in the world that doesn't do DDI and just has
one incoming number?
As I said, if while the caller is in the IVR they dial 101 it works
properly. But some will not know our extension numbers so the IVR rings
several handsets and the first one to pick up gets the call. Why isn't
that information set as the destination EXTEN?
I am beginning to think this is probably a bug. It has nothing to do
with Macros. I have tried without.
Alan
>> [tolc_menu] ; Welcome and information to callers
>> exten => s,1,Answer()
>> exten => s,n,Wait(2)
>> exten => s,n,Background(welcome-to-tolc) ; Say Hello
>> exten => s,n,Wait(1)
>> exten => s,n(tryagain),Background(enter-ext-of-person&or) ; Enter
>> extension number if known, or
>> exten => s,n,Background(pls-stay-on-line) ; Trying to connect...
>> exten => s,n,WaitExten(5)
>> exten => s,n,Macro(belllord,${ALANL}&${ALANB},303)
>>
>> exten => _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
>>
>> exten => _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
>>
>>
>> The Vars ALANL and ALANB are:
>> ALANL=SIP/101
>> ALANB=IAX2/alanb/202
>>
>>
>> Here is the Macro belllord:
>>
>> [macro-belllord]
>> exten => s,1,Dial(${ARG1},20,t)
>> exten => s,n,Goto(s-${DIALSTATUS},1)
>>
>> exten => s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
>> voicemail context, ${ARG2} is the mailbox number to dial
>> exten => s-NOANSWER,n,Hangup()
>>
>> exten => s-BUSY,1,Voicemail(${ar...@business,b)
>> exten => s-BUSY,n,Hangup()
>>
>> exten => _s-.,1,Goto(s-NOANSWER,1)
>>
>>
>> Here is the call-extension Macro:
>>
>> [macro-call_extension]
>> exten => s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
>> exten => s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.
>>
>> exten => s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)
>>
>> exten => s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)
>>
>> exten => _s-.,1,Goto(s-NOANSWER,1)
>>
>>
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I create an IVR/Dial Group that worksproperly?
On 17/07/09 16:29, Adam Robins wrote:
> Have you tried replacing the "s" extension with "_x."?
Thanks, yes I have.
Unfortunately, all that did was to change "s" to the number of our
incoming trunk (i.e. the dialled number). It still does not get set to
the number of the final extension to which the call gets connected.
Cheers
Alan
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord (News)
> Sent: Friday, July 17, 2009 11:12 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] How do I create an IVR/Dial Group that
> worksproperly?
>
> On 17/07/09 14:14, Danny Nicholas wrote:
>> I may 100% off here, but I seem to recall reading in the last 2 days threads
>> that macro dialing messes with CDR entries. I would try replacing one of
>> your macro lines with a straight Dial command to verify this.
>
> Thanks Danny, but that doesn't really help. I have tried moving the
> contents of the offending Macro into the IVR menu itself and using a
> Dial() command. But it makes no difference. The call is still on the "s"
> extension and the CDR records the connection with the correct callerid
> but with the destination as "s". Which is not what I want.
>
> If the caller dials an extension number, say 101, then it all works
> fine. The problem is when trying to automatically dial from within the
> plan it fails. I need to somehow change "s" to the end extension number
> of the person who actually picks up the phone.
>
> I am trying to understand how other people configure their * to achieve
> the requirement I specified below.
>
> I can't believe it is this hard to do. But I fail to see how I can
> achieve it, because there is no extension - other than "s" - when the
> caller enters the dialplan. I want the caller to be automatically
> connected to one or other of our extensions if they do not know the
> extension number to dial themselves.
>
> I guess I am trying to find out if I have set this up totally *wrong*
> and perhaps I should be using a queue or something, but that seems a bit
> overkill...
>
> Alan
>
>
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord
>> (News)
>> Sent: Friday, July 17, 2009 3:23 AM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] How do I create an IVR/Dial Group that
>> worksproperly?
>>
>> Hi all,
>>
>> I am trying to understand how I can get a simple IVR scenario to work
>> properly (having already removed most of my hair...).
>>
>> The basic requirement is as follows:
>>
>> * Caller arrives at our main number
>> * Caller is greeted and then told they can enter an extension number, if
>> known, or wait and their call will be connected to an available rep.
>> * The IVR then dials a group of extensions (if the caller didn't enter
>> one obviously).
>> * Someone picks up the call and the connection is established and logged.
>>
>> Now, I have all of this working apart from the last piece.
>>
>> My IVR rings various extensions and I can pick up the call just fine.
>> But my problem is that the data asterisk records regarding the call is
>> wrong.
>>
>> It correctly identifies the CallerID, but it always records the
>> destination as "s". Not the extension of, for example my SIP phone (101).
>>
>> If the incoming caller dials 101 whilst in the IVR, the log is correct.
>>
>> I can see *why* I am having this problem (There is no extension when you
>> arrive in the IVR other than "s"), but I cannot see *how* to fix it.
>>
>> Please can I ask how do others handle this so it works properly (I've
>> included the basics of my DP below)?
>>
>> I'm running Asterisk 1.4.21.2~dfsg-1ubuntu3 on Ubuntu Server 8.10.
>>
>> Thanks
>>
>> Alan
>>
>>
>> Here is the IVR which callers are dropped into:
>>
>> [tolc_menu] ; Welcome and information to callers
>> exten => s,1,Answer()
>> exten => s,n,Wait(2)
>> exten => s,n,Background(welcome-to-tolc) ; Say Hello
>> exten => s,n,Wait(1)
>> exten => s,n(tryagain),Background(enter-ext-of-person&or) ; Enter
>> extension number if known, or
>> exten => s,n,Background(pls-stay-on-line) ; Trying to connect...
>> exten => s,n,WaitExten(5)
>> exten => s,n,Macro(belllord,${ALANL}&a
Re: [asterisk-users] How do I create an IVR/Dial Group that worksproperly?
On 17/07/09 14:14, Danny Nicholas wrote:
> I may 100% off here, but I seem to recall reading in the last 2 days threads
> that macro dialing messes with CDR entries. I would try replacing one of
> your macro lines with a straight Dial command to verify this.
Thanks Danny, but that doesn't really help. I have tried moving the
contents of the offending Macro into the IVR menu itself and using a
Dial() command. But it makes no difference. The call is still on the "s"
extension and the CDR records the connection with the correct callerid
but with the destination as "s". Which is not what I want.
If the caller dials an extension number, say 101, then it all works
fine. The problem is when trying to automatically dial from within the
plan it fails. I need to somehow change "s" to the end extension number
of the person who actually picks up the phone.
I am trying to understand how other people configure their * to achieve
the requirement I specified below.
I can't believe it is this hard to do. But I fail to see how I can
achieve it, because there is no extension - other than "s" - when the
caller enters the dialplan. I want the caller to be automatically
connected to one or other of our extensions if they do not know the
extension number to dial themselves.
I guess I am trying to find out if I have set this up totally *wrong*
and perhaps I should be using a queue or something, but that seems a bit
overkill...
Alan
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord
> (News)
> Sent: Friday, July 17, 2009 3:23 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] How do I create an IVR/Dial Group that
> worksproperly?
>
> Hi all,
>
> I am trying to understand how I can get a simple IVR scenario to work
> properly (having already removed most of my hair...).
>
> The basic requirement is as follows:
>
> * Caller arrives at our main number
> * Caller is greeted and then told they can enter an extension number, if
> known, or wait and their call will be connected to an available rep.
> * The IVR then dials a group of extensions (if the caller didn't enter
> one obviously).
> * Someone picks up the call and the connection is established and logged.
>
> Now, I have all of this working apart from the last piece.
>
> My IVR rings various extensions and I can pick up the call just fine.
> But my problem is that the data asterisk records regarding the call is
> wrong.
>
> It correctly identifies the CallerID, but it always records the
> destination as "s". Not the extension of, for example my SIP phone (101).
>
> If the incoming caller dials 101 whilst in the IVR, the log is correct.
>
> I can see *why* I am having this problem (There is no extension when you
> arrive in the IVR other than "s"), but I cannot see *how* to fix it.
>
> Please can I ask how do others handle this so it works properly (I've
> included the basics of my DP below)?
>
> I'm running Asterisk 1.4.21.2~dfsg-1ubuntu3 on Ubuntu Server 8.10.
>
> Thanks
>
> Alan
>
>
> Here is the IVR which callers are dropped into:
>
> [tolc_menu] ; Welcome and information to callers
> exten => s,1,Answer()
> exten => s,n,Wait(2)
> exten => s,n,Background(welcome-to-tolc) ; Say Hello
> exten => s,n,Wait(1)
> exten => s,n(tryagain),Background(enter-ext-of-person&or) ; Enter
> extension number if known, or
> exten => s,n,Background(pls-stay-on-line) ; Trying to connect...
> exten => s,n,WaitExten(5)
> exten => s,n,Macro(belllord,${ALANL}&${ALANB},303)
>
> exten => _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
>
> exten => _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
>
>
> The Vars ALANL and ALANB are:
> ALANL=SIP/101
> ALANB=IAX2/alanb/202
>
>
> Here is the Macro belllord:
>
> [macro-belllord]
> exten => s,1,Dial(${ARG1},20,t)
> exten => s,n,Goto(s-${DIALSTATUS},1)
>
> exten => s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
> voicemail context, ${ARG2} is the mailbox number to dial
> exten => s-NOANSWER,n,Hangup()
>
> exten => s-BUSY,1,Voicemail(${ar...@business,b)
> exten => s-BUSY,n,Hangup()
>
> exten => _s-.,1,Goto(s-NOANSWER,1)
>
>
> Here is the call-extension Macro:
>
> [macro-call_extension]
> exten => s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
> exten => s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.
>
> exten => s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)
>
> exten => s-BUSY,1,Voicemail(${macro_ext...@garde
[asterisk-users] How do I create an IVR/Dial Group that works properly?
Hi all,
I am trying to understand how I can get a simple IVR scenario to work
properly (having already removed most of my hair...).
The basic requirement is as follows:
* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if
known, or wait and their call will be connected to an available rep.
* The IVR then dials a group of extensions (if the caller didn't enter
one obviously).
* Someone picks up the call and the connection is established and logged.
Now, I have all of this working apart from the last piece.
My IVR rings various extensions and I can pick up the call just fine.
But my problem is that the data asterisk records regarding the call is
wrong.
It correctly identifies the CallerID, but it always records the
destination as "s". Not the extension of, for example my SIP phone (101).
If the incoming caller dials 101 whilst in the IVR, the log is correct.
I can see *why* I am having this problem (There is no extension when you
arrive in the IVR other than "s"), but I cannot see *how* to fix it.
Please can I ask how do others handle this so it works properly (I've
included the basics of my DP below)?
I'm running Asterisk 1.4.21.2~dfsg-1ubuntu3 on Ubuntu Server 8.10.
Thanks
Alan
Here is the IVR which callers are dropped into:
[tolc_menu] ; Welcome and information to callers
exten => s,1,Answer()
exten => s,n,Wait(2)
exten => s,n,Background(welcome-to-tolc) ; Say Hello
exten => s,n,Wait(1)
exten => s,n(tryagain),Background(enter-ext-of-person&or) ; Enter
extension number if known, or
exten => s,n,Background(pls-stay-on-line) ; Trying to connect...
exten => s,n,WaitExten(5)
exten => s,n,Macro(belllord,${ALANL}&${ALANB},303)
exten => _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
exten => _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
The Vars ALANL and ALANB are:
ALANL=SIP/101
ALANB=IAX2/alanb/202
Here is the Macro belllord:
[macro-belllord]
exten => s,1,Dial(${ARG1},20,t)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
voicemail context, ${ARG2} is the mailbox number to dial
exten => s-NOANSWER,n,Hangup()
exten => s-BUSY,1,Voicemail(${ar...@business,b)
exten => s-BUSY,n,Hangup()
exten => _s-.,1,Goto(s-NOANSWER,1)
Here is the call-extension Macro:
[macro-call_extension]
exten => s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
exten => s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.
exten => s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)
exten => s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)
exten => _s-.,1,Goto(s-NOANSWER,1)
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[asterisk-users] Struggling with Macros and "s" Extension
Hi all,
I'm sure this has been done before but I just can't figure it out.
On my * box I have a simple IVR:
[tolc_menu] ; Welcome and information to callers
exten => s,1,Answer()
exten => s,n,Wait(2)
exten => s,n,Background(welcome-to-tolc) ; Say Hello
exten => s,n,Wait(1)
exten => s,n(tryagain),Background(enter-ext-of-person&or) ; Enter
extension number if known, or
exten => s,n,Background(pls-stay-on-line) ; Trying to connect...
exten => s,n,WaitExten(5)
exten => s,n,Macro(belllord,${ALANL}&${ALANB},303)
exten => _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
exten => _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
.
.
.
Hopefully you'll see that the caller can either enter an extension
number or wait. If they wait, we use macro-belllord:
[macro-belllord]
exten => s,1,Dial(${ARG1},20,t)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
voicemail context, ${ARG2} is the mailbox number to dial
exten => s-NOANSWER,n,Hangup()
exten => s-BUSY,1,Voicemail(${ar...@business,b)
exten => s-BUSY,n,Hangup()
exten => _s-.,1,Goto(s-NOANSWER,1)
The Vars ALANL and ALANB are:
ALANL=SIP/101
ALANB=IAX2/alanb/202
If I call in, dial the extension (say 101) and connect, then the Link
Event on the AMI port (and in CDRs) correctly displays *both* numbers of
the connection.
For that scenario we use macro-call_extention:
[macro-call_extension]
exten => s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
exten => s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.
exten => s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)
exten => s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)
exten => _s-.,1,Goto(s-NOANSWER,1)
If however I wait and let macro-belllord do it's stuff. I only ever see
"s" as the called party's number.
I really need to know what that extension number is.
Could someone help me and show how I can rejig this? It was suggested to
do something with ${MACRO_EXTEN} but I can't get it at all...
Many thanks in advance.
Alan
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Re: [asterisk-users] Setting up a "secure" AMI?
On 09/07/09 14:40, Steve Howes wrote: > On 9 Jul 2009, at 13:05, Alan Lord (News) wrote: >> Reading the page >> http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.conf >> got me a little concerned regarding having an open channel between the >> two machines and there is scant information about setting up a more >> secure connection. >> >> Can anyone offer any good links or howtos for this? > > You can probably tunnel it over SSH. Yes, I am trying to set up a [simple] stunnel connection. Nearly there... If anyone has a decent "how to" I'd live to have a link :-) Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting up a "secure" AMI?
Hi All, I've just upgraded our CRM and it has an Asterisk Integration Module that I would like to test out. The CRM is running on one of our hosted servers in the cloud. The Asterisk server is running in my office. I am running Asterisk 1.4.21.2~dfsg-1ubuntu3. Reading the page http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.conf got me a little concerned regarding having an open channel between the two machines and there is scant information about setting up a more secure connection. Can anyone offer any good links or howtos for this? The CRM is vtiger and I couldn't see any references to ssl in the php code. TIA Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CRMy type app?
On 01/07/09 16:29, Gordon Henderson wrote: > On Wed, 1 Jul 2009, Alan Lord (News) wrote: > >> On 29/06/09 18:26, Gordon Henderson wrote: >>> >>> Looking for a (windows) app. that will listen to the manager interface >>> then pop-up a web browser pointing to a page on an incoming phone call.. >>> >>> Not looking for outlook integration, or outbound dialling, just to >>> recognise an incoming call and poke a URL at a website in a browser and >>> I've absolutely no idea how to do it in the MS windows world... >>> >>> Any clues appreciated.. (More pointing to an existing app. rather than how >>> to write it myself!) >> >> Hi Gordon, >> >> Have you looked at ADM before? It might be suitable... >> >> http://adm.hamnett.org/ > > I saw it - are you part of the team, or if not, then I hope someone from > there is listening in... > > So I saw it, but you know what - the website didn't actually tell me what > it does. I think it's great to have a bloggy/wordpressy/wiki sort of > website, but the front page is lacking a missing "What does ADM do" > paragraph or link to a page... Sure, there's screen shots, documentation, > forums, etc. but if there was a single paragraph at the top that said > exactly what it can do, then I'd have spent more time looking at it.. > > I have now spent some time on the site, but since I've already tested ADAT > and it does what I need, it'll take some persuading to make me change... lol, I'm not anything to do with it. My business partner is using it though. It's a Java app (cross platform) that does the things you were looking for. It does seem to work for him although I haven't tried it myself. I just saw your post and thought it seemed to be a good fit. From an earlier blog post on his site: ADM provides some great features: * Automatic on-call volume reduction * One click dial from clipboard (paste number onto tray icon) * Integrated phonebook * List/Redirect/Hangup all active calls * One click call forward setup * Bluetooth presence detection to redirect calls when you walk out of the office * Pop up browser on incoming call (integrate with your CRM to auto load customers details when they call) * Cisco phone integration (auto speakerphone) * Slide-in popup on incoming call, with Answer(cisco only), Hold, Busy and Redirect buttons , CallerID and duration Cheers Al PS - he's a Brit too. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CRMy type app?
On 29/06/09 18:26, Gordon Henderson wrote: > > Looking for a (windows) app. that will listen to the manager interface > then pop-up a web browser pointing to a page on an incoming phone call.. > > Not looking for outlook integration, or outbound dialling, just to > recognise an incoming call and poke a URL at a website in a browser and > I've absolutely no idea how to do it in the MS windows world... > > Any clues appreciated.. (More pointing to an existing app. rather than how > to write it myself!) Hi Gordon, Have you looked at ADM before? It might be suitable... http://adm.hamnett.org/ Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Learn Asterisk
On 22/06/09 18:20, David @ULC wrote: > > What the best website and book to start learning asterisk ? Website: Google, http://www.voip-info.org Book: TFOT (The future of Telephony) Google for it , it is freely/legally downloadable. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP phone recommendation
On 03/06/09 08:37, Rilawich Ango wrote: > Hi all, >Any good recommendation of IP phone in term of sound quality and > price (reasonable) using with asterisk? > ango Not sure where you are in the world, or what you really need but I like the Siemens Gigaset IP DECT phones. The S685IP is really nice: http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/ HTH Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Atcom] Asterisk + LAMP on 128MB RAM?
On 01/06/09 10:27, Vincent wrote: > Hello > > I'm thinking of selling an Asterisk server based on Atcom's IP02 > solid-state unit with one FXO and one FXS ports: > > http://atcom.cn/En_products_IP02.htm > > By default, this unit based on a 400MHz Blackfin 532 chip only has > 64MB RAM and 256MB of NAND flash. Those can be increased to 128MB and > 1GB, respectively. > > Do you think I can install Linux + Asterisk + LAMP (replacing MySQL > with Firebird, to avoid license costs) on the default specs, or will > it be a bit short? Check out the Astfin project (http://blog.astfin.org/?page_id=2). I'm guessing they have already done what you need... Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
On 24/05/09 19:21, Gordon Henderson wrote: > Here's an example: > > http://www.wppltd.demon.co.uk/WPP/Wiring/UK_telephone/uk_telephone.html > > In these enlightened days, it's normal to not use the 3rd wire internally > on extension as it has been known to degrade an ADSL signal. I think this last comment is very true indeed. I purchased something called an i-plate here in the UK. I'd seen many feedbacks about how it had improved broadband performance. It is a basically a small plugin device for the BT master socket that contains a decent filter and a terminator for the bell-wire. They cost around £10 and take only a minute to fit. Here's what happened when I plugged it in: http://www.theopensourcerer.com/2009/03/31/broadband-magic-with-the-i-plate/ And performance has even improved slightly more since that post too. Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Building a System.
On 11/05/09 04:21, John F. Ervin wrote: > Are there (??) instructions for people who are experienced at the > Trixbox level but wish to move on? Sure, the TFOT book is a great start. If you want to use Ubuntu or Debian rather than Centos then Asterisk is in the Debian and Ubuntu Server Repositories. # sudo apt-get install asterisk # sudo m-a -f get zaptel-source # sudo ECHO_CAN_NAME=OSLEC m-a -t a-i zaptel (The last command might not be needed any more as I believe OSLEC is now the default EC) Should do most of it. I written a few blog articles on Asterisk (building from scratch and installing on Ubuntu etc. Including Zaptel and OSLEC for the x100p) Here's one for getting it running on Ubuntu server: http://www.theopensourcerer.com/2009/02/12/asterisk-zaptel-oslec-and-ubuntu-server/ And here's all of them: http://www.theopensourcerer.com/tag/asterisk/ There are, of course, many more guides and advice out there too. Google is your friend as it the extremely useful http://www.voip-info.org/ wiki.. HTH Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions on X100P/X101P cards
On 06/05/09 13:43, Vincent wrote: > Hello, > > I'm looking for a dirt cheap solution for SOHO use to handle at most > a couple of POTS lines, and I notice that X10?P cards go for $15 on > eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma. > > I have a couple of questions about those cheap FXO cards: > > 1. Are they all glorified softmodems, ie. none has an on-board CPU or > DSP and outsources all processing to the computer's CPU? > > 2. Are they all bad, no matter what chipset is used (Intel, Motoral, > Ambient)? If not, which offer good enough quality to handle a single > POTS line? > > 3. Why are they often bad quality? Because the driver itself is badly > written? Because PC's don't have enough speed to handle the tasks > using their own CPU (hard to believe, but I don't know)? Hi Vincent, I bought a cheap "eBay" X100p card over a year ago. When I first tried it was appalling. I couldn't get rid of the echo and noise no matter what. I then came across OSLEC (at the time a new Free Echo Canceller). A bit of hacking to get it to work and hey-presto! No more echo. I have been using the same card ever since with no noticeable issues. I think OSLEC is now the default EC for many distributions so I would have thought you will be fine although, of course, YMMV. For a cheap backup to your VOIP service they do the job. I wouldn't use them for a "proper" system though. HTH Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compact, fanless appliance?
On 06/05/09 08:28, Gordon Henderson wrote: > One little tip: You need to compile Asterisk for an i586 processor as > the VIA processor is missing a few (mmx, etc.) instructions that a full > blown i686 has. Hi Gordon, I'm using a VIA C7 on a Jetway board (http://linitx.com/viewproduct.php?prodid=11212). This is my cpuinfo. Isn't that an i686 class? cat /proc/cpuinfo processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 10 model name : VIA Esther processor 1200MHz stepping: 9 cpu MHz : 1099.969 cache size : 128 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge cmov pat clflush acpi mmx fxsr sse sse2 tm nx up pni est tm2 rng rng_en ace ace_en ace2 ace2_en phe phe_en pmm pmm_en bogomips: 2199.93 clflush size: 64 power management: Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI PHP
On 04/05/09 21:17, James A. Shigley wrote: > I’m just trying to make a real simple Survey via php. Just want it to > play the Question Files, wait for a response, save the response into the > correct variable and then email it all. Packt sent me a book to review recently: Asterisk AGI Programming. My write up is here (http://www.theopensourcerer.com/2009/04/16/asterisk-agi-programming-with-packt/) if you are interested, it might be a big help depending on your time-scales etc. Al PS - Disclaimer: Packt sent me this book for free and under no obligation. The review is genuine and unbiased. We are in Packt's affiliate scheme too, so we do get a few cents if you decide to buy the book via a link on my blog. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI Programming
Hi all, This isn't meant to be spam I thought some of you might find it interesting. Packt Publishing approached me a few weeks ago and asked if I would like to review a book or two for them on my blog. The first one they sent me is called Asterisk Gateway Interface Programming and has only just been released. It was written by Nir Simionovich. You can read my review here: http://www.theopensourcerer.com/2009/04/16/asterisk-agi-programming-with-packt/ They have also sent me a book on Trixbox CE 2.6 which I will get round to reading and reviewing over the coming month or so. Cheers Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Siemens Gigaset Phones get mute function.
Hi, I know this is a little OT but there are many Asterisk users of the excellent Siemens DECT/VOIP phones like the S685IP and 475IP and this is probably newsworthy for them. One of the biggest bug bears has been no mute function on the handset. When I woke up this morning, the handset told me there was a firmware update. I updated and then visited the web site to find out what had been fixed (quite a lot of new features have been added): http://gigaset.com/shc/0,1935,hq_en_0_152411_rArNrNrNrN_variation%253A-5_pageType%253Adownloads_imagePos%253A0,00.html#content Gigaset C470 IP / C475 IP / S675 IP / S685 IP Firmware update 04/2009 Download version: 02184 New features: * E-mail viewer (with C47H, S45, S67H, S68H handsets) * Mute function. Turn off the handset's microphone during an external call with the left display key. * Send and receive SMS messages via VoIP* * VoIP: If the telephone cannot establish a VoIP connection, it automatically dials via the fixed line network (auto-fallback to PSTN). * VoIP: Call transfer via R key * VoIP: An incoming call indicated in parallel at different VoIP devices (parallel ringing) will not be stored in the "Missed calls" list if the call was accepted at one of the devices.* * Online directory: display the postal codes in search results.* * Online directory: when starting a new search the cities used in the last searches will be displayed. * Fixed line access codes can be stored in the phone. * Some languages will be loaded on to the base via the internet, depending on the language set on the handset. * Extended RTP port range (1024-55000) * DHCP Option 114 implemented.* * DHCP Option 120 implemented.* * Web Configurator: option to specify whether the "area code" is dialled as well. * Web Configurator: display RTP port range * Web Configurator: new languages - Arabic and Russian * Web Configurator: enhanced PIN protection - warning if the default pin () has not been changed. Improvements: * The automatic search function for firmware updates is enabled even if the internet connection is temporarily interrupted during the night. * The indication "Anonymous call activated" will no longer be displayed in the idle mode of the handset. * The country code is synchronised between base station and handset. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cant get a x100p works
Manolet Gmail wrote: > I have an ubuntu 8.10 machine. Linux 2.6.27-14-generic > > i want to configure a x100p card an use it with asterisk, so i download, > compile and install: > > asterisk-1.4.24 > dahdi-linux-2.1.0.4 > dahdi-tools-2.1.0.2 > libpri-1.4.9 > > i try almost everything i found on the net but without success: I have a working X100p on Ubuntu 8.10 server using zaptel and oslec (I seriously recommend you use oslec rather than MG2). I blogged about it here: http://www.theopensourcerer.com/2009/02/12/asterisk-zaptel-oslec-and-ubuntu-server/ I must clean that post up a bit; it looks a mess, but basically ignore all the strikeout text. Thanks to Tzafir who helped me simplify it quite a bit. HTH Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Zaptel] Why no driver for PCI voice modems?
Fred wrote: > Hello > > Considering how cheap PCI modems are compared to even the cheapest > PCI hardware from Digium, OpenVox, Sangoma, etc I was wondering > why Zaptel can't be used with those to connect an Asterisk server to > a POTS line for low-level use? It just seems overkill for SOHO users > who only get a few calls a day. Hi Fred, just purchase an X100[p] clone on ebay. I bought one last year from a seller in the USA and it cost me about £17 (GBP) including shipping. Using Zaptel and OSLEC it is absolutely fine. HTH Alan PS - I did a quick search and here is the type of thing I mean. I have no idea of the seller or this particular board but you get the idea I'm sure: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=220380070996 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Zaptel] Why no driver for PCI voice modems?
Fred wrote: > Hello > > Considering how cheap PCI modems are compared to even the cheapest > PCI hardware from Digium, OpenVox, Sangoma, etc I was wondering > why Zaptel can't be used with those to connect an Asterisk server to > a POTS line for low-level use? It just seems overkill for SOHO users > who only get a few calls a day. Hi Fred, just purchase an X100[p] clone on ebay. I bought one last year from a seller in the USA and it cost me about £17 (GBP) including shipping. Using Zaptel and OSLEC it is absolutely fine. HTH Alan PS - I did a quick search and here is the type of thing I mean. I have no idea of the seller or this particular board but you get the idea I'm sure: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=220380070996 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need to find small footprint asterisk platform
Anthony Plack wrote: > Hey all, > I have a potential project which calls for a very small form-factor computer > like this: > > http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp > > However, I am needing an FXS port integrated into a small footprint computer. > Nothing larger than a WiFi router or gateway device, but the smaller the > better, and able to run Asterisk with at least a spare USB port and > preferably WiFi on the system (but no necessary). This is a pretty cool product: http://www.rowetel.com/ucasterisk/store.html The hardware design is Open Source too ;-) Cheers Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outlook integration?
Paul Hales wrote: > Noojeeclick? > > http://www.noojee.com.au/Page/NoojeeClick Thanks for that. Not heard of NoojeeClick before. Their site is not responding right now but the Firefox add-on page is up. when I get chance I will try it out. https://addons.mozilla.org/en-US/firefox/addon/8510 I was aware of ADM. Thanks again, Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outlook integration?
Dean Collins wrote: > ADA Forums: http://forums.digium.com/index.php?c=8 > ADA Download: http://dl1.digium.com/ADA/ADAInstall.exe > ADA Administrators Guide: http://dl1.digium.com/ADA1.1/ADA_Admin_Manual.pdf Thanks for the links. I hadn't seen that before. The product is "kind of" interesting, but does anyone know of something similar for non-windows desktops? Thanks Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OSLEC not being loaded on Ubuntu Intrepid
Tzafrir Cohen wrote: >> But when the wcfxo module is loaded, it is not loading the oslec module. >> There is an oslec.ko in /lib/modules/my-running-kernel-version/misc/oslec/ >> >> According to launchpad, oslec should be the default ec now for zaptel. >> >> Anyone got any ideas please? > > http://bugs.debian.org/510858 > > Fixed in SVN: http://svn.debian.org/viewsvn/pkg-voip?rev=6684&view=rev > > As mentioned there, the workaround is to set ECHO_CANC_NAME explicitly: > > ECHO_CAN_NAME=oslec m-a a-i zaptel > Thanks Tzafrir. That's a much easier workaround :-) I did manage to sort it out, but I manually edited zconfig.h and rebuilt the tarball before running the m-a again. It worked but was a bit of a PITA. BR Alan Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OSLEC not being loaded on Ubuntu Intrepid [SOLVED]
Alan Lord (News) wrote:
> I wonder if anyone has any ideas on this.
>
> I have recently migrated my server from a custom built Linux with
> Asterisk, Zaptel and Oslec running fine, to Ubuntu Server 8.10.
>
> I have Asterisk installed via synaptic at it works fine.
>
> I have built and installed the zaptel package by doing the following
> commands:
>
> sudo m-a -t build zaptel
> cd /usr/src
> sudo dpkg -i zaptel-modules-{version}.deb
> sudo modprobe zaptel
> sudo modprobe wcfxo
>
> But when the wcfxo module is loaded, it is not loading the oslec module.
> There is an oslec.ko in /lib/modules/my-running-kernel-version/misc/oslec/
>
> According to launchpad, oslec should be the default ec now for zaptel.
Answering my own question - the file zconfig.h is still declaring MG2 as
the default ec.
I edited this file to define OSLEC instead, zipped up the archive and
then rebuilt the zaptel module as above. Now works.
Maybe this will help someone else too.
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[asterisk-users] OSLEC not being loaded on Ubuntu Intrepid
I wonder if anyone has any ideas on this.
I have recently migrated my server from a custom built Linux with
Asterisk, Zaptel and Oslec running fine, to Ubuntu Server 8.10.
I have Asterisk installed via synaptic at it works fine.
I have built and installed the zaptel package by doing the following
commands:
sudo m-a -t build zaptel
cd /usr/src
sudo dpkg -i zaptel-modules-{version}.deb
sudo modprobe zaptel
sudo modprobe wcfxo
But when the wcfxo module is loaded, it is not loading the oslec module.
There is an oslec.ko in /lib/modules/my-running-kernel-version/misc/oslec/
According to launchpad, oslec should be the default ec now for zaptel.
Anyone got any ideas please?
Thanks
Alan
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Re: [asterisk-users] What do you use? .conf or AEL?
That was quite an interesting set of responses. I didn't get any impression that there is a strong preference either way. Thanks for all the replies. Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What do you use? .conf or AEL?
Hi all, I built my first asterisk using the traditional (?) .conf files and constructs. I recall reading books at the time about AEL but it seemed "new" and untested so I left it alone. Now, I'm interested to poll the audience here to see if I should look into using AEL instead (or in addition to) for future work. TIA ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk work with a dynamic IP?
Ronald Wiplinger (Lists) wrote: > I know I can setup asterisk without Internet at all and it works as > local pbx. > > Would an asterisk box work with a dynamic IP, with a dyndns name? > What must I take care if I try that? I had my * server behind my adsl router that was getting a dynamic Ip address. I simply created a domain for my site at http://www.dyndns.com/ (free) and it worked fine. Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] echo cancellation for sip phones
Pezhman Lali wrote: > Dear, > the sip phones that registered, in to the asterisk 1.4.x have the echo > in their callings to pstn. > how this echo can be canceled? H - you don't give much to go on... What is the connection to the PSTN (i.e. what kind of card, interface etc...) The echo is almost certainly coming from that area rather than the SIP phones themselves. If you have an analogue PSTN card without h/w echo cancellation, I would suggest trying the OSLEC echo canceller. This is[was?] not installed by default with the zaptel drivers. HTH Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for a good lightweight Linux softPhone
Cory Andrews wrote: > http://blog.voipsupply.com/new-products/free-sip-softphone-roundup > Good roundup, thanks. Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)
Olivier wrote: > Alan, > > Did you get any success with MWI ? > With mine, Asterisk is getting 481 replies whenever Asterisk sends > NOTIFY updates. > > Cheers I don't think so no. The lamp blinks when I've missed a call but I don't think it correctly identifies if there are messages in the * vmailbox. To be honest it hasn't troubled me that much as when a message is left on the * server I get an email notification too so it's not like I miss much :-). But it would be good to try and fix this. I get a pair of error messages like this frequently (about 1/minute I guess) from the * server: [Oct 30 08:42:05] WARNING[14051]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. [Oct 30 08:42:15] WARNING[14051]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. which I believe might have something to do with it? 10.0.0.2 is the Asterisk server. My sip.conf is like so: > [general] > srvlookup=yes > disallow=all > allow=alaw > allow=g722 > allow=gsm > dtmfmode=auto > subscribemwi=yes > > [101] > type=friend > callerid=Alan Lord <101> > secret=bigsecret > qualify=yes ; Qualify peer is no more than 2000 ms away > nat=no ; This phone is not natted > host=dynamic ; This device registers with us > canreinvite=no ; Asterisk by default tries to redirect > context=alanl ; the internal context controls what we can do > mailbox=101 ; Voicemail Boxes The S685IP supports G722 which we have been testing between our offices (sounds great!). If anyone has any ideas that would be cool. Cheers Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)
Olivier wrote: > I'll reply to the correct thread > > [featuremap] > blindxfer => ## ; Blind transfer > ;disconnect => *0 ; Disconnect > ;automon => *1 ; One Touch Record > atxfer => A ; Attended transfer > > so set atxfer to 'A' and DTMF relay Application signal on the Gigaset to > 'A' (without quotes) > > and transfer works as expected > > Robb > > Thanks for replying ! > I'll give it a try and report to the list I just tested this and it seems to work with the Siemens S685IPs. This thread was such a coincidence. We were trying to get attended transfer to work last night but setting the atxfer to "normal" things like *2 just didn't work. I just set my S685IP base station to A for the Application Signal and set A in the features.conf and behold, when I now press the R key, Attended Transfer :-) Thanks Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone using an Intel Atom ?
Gordon Henderson wrote: > Damn - I've just found it in the UK too: > >http://www.lambda-tek.com/componentshop/index.pl?prodID=1606310 > > Must resisst . > > I just wish there was a fanless version - one feature which I like in the > VIA boards I use. Wow, that's an amazing price for the mobo. Though, like you, WTF do Intel insist on using a chipset that needs fan cooling and draws about 4 times as much power as the processor? It's soo stupid. I *can* resist until they sort out a decent chipset. Or until VIA ship a dual core Nano. ;-) It's been an interesting thread though. I love Steve's idea of using a netbook for Asterisk demos - that's really cool. Cheers Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Advice on ISDN and Asterisk in the UK
Gordon Henderson wrote: > On Fri, 24 Oct 2008, Alan Lord wrote: >> I used to have an ISDN-2 line into my home office. BT wrote to me about >> 2 years ago and said they were discontinuing the service. They converted >> my dual channel BRI back into a single POTS. > > Sure it was ISDN2e and not Home or Business Highway? They killed off an > the HH and BG lines some time back and converted them back to POTS. I've > no idea why - I'd cancelled my HH line some time before the cut-off date. Yes, I think it was called Home Highway (but it *is* just a BRI at the end of the day). It had a USB port on the NTE which I never really grokked. Not being a Windows user and all. >> I built a little Asterisk server, stuck an X100p in it for backup calls >> should my broadband go down (on a separate POTS line) and got two >> non-geo 0844 IAX trunks for free instead. >> >> Who lost out there then? > > Well, quite. BT have their good points, but also their stupidly bad > points too. > > They phone me up once a month at present and ask me why I'm not placing > any outgoing calls with them. When I try to tell them why, (because I > run my own phone company!) because my reply is not in the script, they > just hang up on me. LOL - That's a good one. Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Advice on ISDN and Asterisk in the UK
Phil Knighton wrote: > Hello all > > What I'm looking for is some plain speaking advice on ISDN. > > Currently using 4 analog lines connecting via a four port TDM400P FXO > card. We need to physically move our installations, and on requesting > the analog lines be moved - our telco (BT) is suggesting we replace our > analog lines with ISDN2. We would have 3 x ISDN2 connections, giving us > six voice channels. They've even offered us free installation of the > lines (as opposed to a £560 charge for moving the analog lines!) Wow, you are lucky. I used to have an ISDN-2 line into my home office. BT wrote to me about 2 years ago and said they were discontinuing the service. They converted my dual channel BRI back into a single POTS. I built a little Asterisk server, stuck an X100p in it for backup calls should my broadband go down (on a separate POTS line) and got two non-geo 0844 IAX trunks for free instead. Who lost out there then? Cheers Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.4.1 error cross compile
satish patel wrote: > I have set env on shell > > KVERS=2.6.22.5 > KSRC=/path/to/kernel-2.6.22.5/source > Maybe I misunderstood you then. I thought you said that your ARM system was using a 2.6.18 kernel? If that is the case, then surely you need to build your zaptel module against that and *not* the kernel on your cross-compiling host machine. But probably I misunderstood what you meant. Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.4.1 error cross compile
Satish Patel wrote: >>> I am using cross compile so i can't update GCC other wise it will effect on >>> my other packages anyway... tell me one thing i have host system kernel >>> version is 2.6.18 and i am compiling ARM embedded rootbuild with other >>> kernel version 2.6.22 so i need to compile my zaptel package with 2.6.22 >>> kernel caz i will use it on target ARM hardware ( IXP425 ). I am doing that >>> but after porting rootfs on target host and when i run "insmod zaptel" >>> command on target board i got error > clfs:/mnt/clfs$ file lib/modules/2.6.22.6/misc/zaptel.ko > lib/modules/2.6.22.6/misc/zaptel.ko: ELF 32-bit MSB relocatable, ARM, > version 1, not stripped > > i dont know why this error coming i think its caz confusion between > host kernel and target kernel I think you may be right. Can you not extract a set of kernel headers for 2.6.18 and point the zaptel build to them when you are making it in your cross-compile environment? I can't remember the switch off hand but I am sure there is a way to point the make scripts at whatever headers you wish. HTH Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Creating Asterisk Binary Package
Steven Howes wrote: > Just copy the src folder and do `make install` on each machine? > Then tar and copy the /etc/asterisk folder if config is important too. > > On 29 Sep 2008, at 08:41, Jim Boykin wrote: > >> Is there a script to create an Asterisk binary package after it is >> compiled on one system. >> >> We do not want to compile Asterisk of each system where we want to >> run. I am sure there is a way but I could find it. >> Another way is after running "./configure --prefix=/your_prefered_layout" and make, when running the "make install" command set the DESTDIR prefix to something like ~/asterisk and it will install everything under that prefix. e.g $ make DESTDIR=$HOME/asterisk install You can then make a tarball of the hierarchy from within the DESTDIR root and extract it into the right place (i.e. "/") of any other host. Of course all this assumes: 1. you know what you are doing ;-) 2. your hosts are all using the same versions of kernels/libraries etc... HTH Alan >> Thanks >> Jim >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: more on Free World Dialup groups and FWDLive
Gordon Henderson wrote: > On Tue, 23 Sep 2008, Steve Totaro wrote: > >> FYI >> >> It looks like FWD is looking for value added service ideas for free as >> a volunteer. > > I got this too - looks like a bit of a mass mailling! And me! And I haven't visited their site, or connected to their servers as IAX2 never worked, for well over a year either. Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reliable wireless SIP phones
Jaap Winius wrote: > Hi list, > > Are there any reliable wireless SIP phones available on the market yet? > > > Since the firmware seems to be the same, there's no way I'm going to > upgrade to the 685IP. I was thinking of trying out the Snom M3, but > according to voip-info.org, that model suffers from similar > reliability problems. > I know it won't help you much but we have two sets of S685IPs (three h/s in each set) and no problems to speak of. I also liked the M3 but I wanted a PSTN port on the b/station too :-( Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which rackable case for mini-ITX boards ?
Joseph L. Casale wrote: > Al, > What did you finally settle on as a firewall for this project? > jlc > Ahhh - good question; I was waiting for someone to ask... I haven't. I only really needed a content filter for the younger members of my family. When I upgraded "the boys'" computer to Edubuntu, I found a great thread that helped me add Dan's Guardian and a proxy onto that machine. (It runs as root and he can't do anything about it... Hee Heee. Until he learns how to hack I guess. But by then I doubt I'll care much about him looking at porn anyway). As for a firewall, we have one built into my dsl router and it's quite locked down. I have only one M$ PC left in the household (but not for much longer) so other forms of protection are largely unnecessary today. Of course that may change over time but for now I'm quite happy. I should think a GUI like Firestarter (http://www.fs-security.com/) would be my next port of call if/when I decide to add a firewall to my server. HTH and thanks for asking. Al PS - I did test Untangle on a VM and on a separate partition of my VIA host as it now installs cleanly on Ubuntu servers using apt. But is was just too much hassle to migrate the existing apps over to an Ubuntu Server. If my OS hadn't been a roll-your-own I would have probably used it. -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which rackable case for mini-ITX boards ?
--[ UxBoD ]-- wrote: > From what people have said Asterisk does not require a huge amount of memory > or CPU then ? I only have a couple of extensions. Running the G729 codec and > will look at the Octava software for the PSTN to reduce echo. > > Regards, > It doesn't seem so to me. For a home office scenario at least. I have a VIA CN700 that I have downclocked to 1Ghz, it has 1G of RAM (A single stick of DDR2) but it runs *a lot of stuff* as well as Asterisk and I have not experienced any issues. The OS is a custom built Linux based on Linux from Scratch (it enables you to build only what you need and compile for the particular hardware of your choice). On this little box I have Asterisk, Tomcat (running the Chandler Calendar Server, and some apps that we use for testing such as Open Bravo and Concourse Suite CRM), Apache, MySQL and PostgreSQL, Samba (acts as a file server for our home network), and other stuff I have probably forgotten. I measure uptime in months and the usual reason for a restart is due to me playing (breaking) or power failures. My home telephony network consists of a trio of Siemens S685IP SIP/DECT phones, two IAX trunks (one to my business partner's home office Asterisk and one to my telephony provider, and a single X100P for a PSTN backup line should the IP services die (I'd recommend using OSLEC for this kind of card - the EC is excellent). This is the Mobo I chose: http://linitx.com/viewproduct.php?prodid=11212 You can read more on my blog (in my sig below) by clicking on the Asterisk tag for example. Cheers Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens Gigaset IP in USA (S685 IP in particular)
[EMAIL PROTECTED] wrote: > That's the purely technological answer, which is completely correct. > > There's a business side to it as well. Siemens is simply not in the > consumer electronics business in North America. They make this decision > consciously. > That's a shame. It's a cracking phone... :-) And one of the busiest links on my blog! -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running asterisk as non root user
Shaun Wingrin wrote: > Hi, > > I've followed instructions of the book "AsteriskFutureOf > TelephonySecEdit" on page 295 onwards ) Link to the Asterisk book: > http://downloads.oreilly.com/books/9780596510480.pdf) and > <http://downloads.oreilly.com/books/9780596510480.pdf) and> get an error > when running service asterisk start. The error is: cat: > /var/run/asterisk.pid: No such file or directory . That sounds like it's probably a permissions thing. I wrote up a "howto" Asterisk as non-root on my blog here: http://www.theopensourcerer.com/2007/10/30/untangle-asterisk-pbx-and-file-server-all-in-one-part-7/ Make sure you check the init script and the permissions of the various directories that asterisk needs read/write too. HTH Alan -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I used to use an Asterisk server, but now it is overkill, ...
Ronald Wiplinger wrote: > I had installed in the office an Asterisk server, but the company is > gone and I could keep the server. > > However, for my family with three members and two phone lines this > server is overkill. I am looking for a compact solution, which is more > suitable for me. > > I want a small & silent box, which can connect two phone lines and 6 > internal VoIP phones and about 6 external VoIP phones. > I would like to have: > 1. Announcements for callers (dial the extension number) > 2. voice mail with mail forwarding > 3. wakeup call > 4. pickup group > 5. call forwarding after 20 seconds, ... > 6. ISN support, Sipbroker support > 7. remote gateway support > > I guess that is all what I would need at home. Hi Ronald, I built my own small low-power server that runs Linux and provides a host of services for our home and our home businesses. Asterisk is just one of the functions and it runs very happily (well, the box has *never* stopped or needed rebooting apart from when I wanted to change something). The VIA C7 board I bought runs at about 7W, has no fan and I have even downclocked it from 1.2Ghz to 1Ghz. I have written some articles on my blog about it, here's the first article: http://www.theopensourcerer.com/2007/09/08/untangle-asterisk-pbx-and-file-server-all-in-one/ For the other instalments use the tag cloud and Asterisk. With the new Atom processor you might get even better power consumption although I have read somewhere that the associated chipsets for the Atom are very thirsty (+20W)... Hope this helps. Alan -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange beep during calls
Felippe Silvestre wrote: > Hi all, > > Our users are complaining about beeps that happen in the middle of some > calls. They are similar to the sound heard you are in a call and press > any button in your phone. Please find bellow some examples of these > beeps(the recordings are in Portuguese, but the beeps are easy to identify): > > http://www.katizak.locaweb.com.br/asterisk/beep.mp3 > http://www.katizak.locaweb.com.br/asterisk/beep2.mp3 > http://www.katizak.locaweb.com.br/asterisk/beep3.mp3 > http://www.katizak.locaweb.com.br/asterisk/beep4.mp3 > > We are sure that our users are not pressing any button in the softphones > during the conversations. > Do you guys are able to identify where these beeps are coming from? > Maybe an * functionality that we need to turn off... We are using > Asterisk 1.4.21.2. > > Thanks. > There was a short discussion on the OSLEC mailing list very recently about something that sounds (forgive the pun) similar. (Sorry I can't add anything else but I deleted them.) I can't recall what the suggestions were but I think someone mentioned possible hardware faults on an analogue line card... Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme
Steve Totaro wrote: > On Mon, Jul 7, 2008 at 6:58 AM, FaberK <[EMAIL PROTECTED]> wrote: >> Hi folks, >> we use meetme application with pin so when a customer joins he's >> prompted for his name. >> Then the voice say:"press one to accept the recording..." >> My question is, is it possible to cut off that request to"press one"? >> Audacity? -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/IAX2 Provider with fallback dialing?
Steve Finkelstein wrote: > Hi all, > > I was curious if anyone can recommend a company that would work with > small businesses, and capable of using a fallback number (mobile > phone, home number etc) in the event SIP or IAX2 peering was to > terminate because of some outage. This could be useful when you do > not have a backup T1 PRI, etc. > > Thanks all, > Here in the UK we use a small company who do exactly that, Pounbury Systems Ltd (http://www.poundbury.com/). We have two VOIP non-geo numbers delivered using IAX2, and they will auto redirect to our home phone or any number you wish should the IP service fail. Good price, quick and friendly... -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird NAT issue ...
sorry for not replying to this sooner! but the canreinvite=no trick worked a treat. thank you -- Alan Williamson Try our free registrationless email/sms reminder http://yourli.st/ b: http://alan.blog-city.com/ Steve Davies wrote: > If the two phones attempt to refer to each other using their external > (NAT) IP addresses rather that their internal addresses, then it will > all go horribly wrong. You do not provide enough information about > asterisk IP addresses or firewalls for a possible solution, but > assuming you are using SIP and asterisk, you could try > "canreinvite=no" against the 2 phones to see if keeping the Asterisk > server in-the-loop helps. > > Also look on the VoIP wiki for "externip" and "localnet" in the > sip.conf configuration. > > Regards, > Steve > > On Mon, Mar 17, 2008 at 1:59 PM, Alan Williamson <[EMAIL PROTECTED]> wrote: >> Afternoon one and all. >> >> I am having some interesting fun with our Asterisk setup. >> >> We have two CISCO handsets (7960) sitting on the same network (NAT). >> >> Each phone can successfully originate calls. >> >> Each phone can be called successfully from outside >> >> Each phone can be directly called by other extensions OUTSIDE the network >> >> >> HOWEVER -- when those 2 phones try to call each other; the connection is >> made, but no voice is heard. >> >> Any advice as to where i need to look? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Asterisk: Install Asterisk as non-root
James Sneeringer wrote: >> Also note that asterisk.conf options override command-line options (and >> not the other way around, as you might have learned to expect from most >> other applications). > > Some asterisk.conf options, such as runuser and rungroup, don't appear > to work at all. I can get Asterisk to run non-root using -U and -G on > the command line, but attempting to do it in asterisk.conf instead > doesn't work for me. The command line is good enough for me, so I > haven't taken the time to figure out why it doesn't work. My uneducated guess would be that for Asterisk to parse the asterisk.conf file it has to be running... Therefore it must already be running as the user which it was told to run as. Alan -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not hearing first prompts
Sherwood McGowan wrote: >> > Hrm...I have encountered this before and sometimes doing an explicit > Answer() then a Wait(2), then calling the service can help. > > Hope this is helpful > > Sherwood McGowan > Bingo! Thanks a bunch. That sorted it. Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not hearing first prompts
Stefan Guenther wrote: > Alan Lord wrote: > > >When I connect to various asterisk services such as VoicemailMain(), > >MeetMe() as examples, I do not get to hear the first greeting messages. > > > >I've tried adding a Wait(1) before or after the application but this > >seems to have no effect. > > > >Is there another setting/parameter I can play with to delay the start > >of the playback of these messages? > > > do you use SNOM phones? > > We had the same problem with a number of SNOM phones. > Unfortunately not. It is the same if we use our Siemens DECT/SIP handsets or the Ekiga softphone... I recall having this problem once before and that it "went away" when I changed from Ekiga to Twinkle. When I get chance, I will re-install Twinkle and see if that exhibits the same problem. Thanks anyway. Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Not hearing first prompts
When I connect to various asterisk services such as VoicemailMain(), MeetMe() as examples, I do not get to hear the first greeting messages. I've tried adding a Wait(1) before or after the application but this seems to have no effect. Is there another setting/parameter I can play with to delay the start of the playback of these messages? TIA Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Asterisk: Install Asterisk as non-root
Lee, John (Sydney) wrote: > I was following the instruction on > http://www.voip-info.org/wiki-Asterisk+non-root to re-install my > Asterisk as non-root when I had the following questions/issues: > > 1) " Use your system's preferred method of adding a new user. Examples: > Red Hat: adduser -c "Asterisk PBX" -d /var/lib/asterisk -u 5060 > asterisk" > ###Why did we have to choose uid as 5060? > ###In fact, do you need to specify the uid at all? Nope - the UID doesn't matter, but it is general practice to keep system (application) UIDs below 100 or 1000 and "normal" users above. So I'd use a number below 100 or 1000 depending on your linux distro's standard. > > 2) "Edit your Asterisk config file (/etc/asterisk/asterisk.conf): > astrundir => /var/run/asterisk > Recompile and reinstall Asterisk." > ### Seems a bit strange to modify this before you recompile. > ### As it turns out, the reinstall did not change the astrundir variable > ### You have to manually modify it if this modification is actually > required. > That won't affect compilation whatsoever. > > 3) "Also, make note that if you're running udev on your system > (linux-2.6), the /dev directory is dynamically populated with device > nodes, meaning that any permissions you set on /dev/zap will be lost on > your next reboot, and you may get a nasty message such as "Asterisk > ended with exit status 1" > when trying to start asterisk. Read the file > /path/to/zaptel-src-1.2.x/README.udev for instructions on how to change > the user/group assigned to /dev/zap. " > ### There is actually no README.udev file in zaptel source. > ### Do I need to worry about this if "uname -r" returns 2.6.18-8.el5 > ### What actually is udev? > udev help linux to dynamically create/remove the interfaces to various hardware devices and so forth. After installing the zaptel module you'll see a udev rules file "zaptel.rules" in your etc/udev configuration area. It doesn't take a genius to work out if or how you need to change anything in that file... > 4) "Asterisk needs read permission for these directories and their > contents: > /etc/asterisk. > chown --recursive root:asterisk /etc/asterisk" > ### root is not in group asterisk > ### All the while, the instruction has been saying to create a user > asterisk > ### under group asterisk. > ### Does it mean to put root into group asterisk as well??? > ### Or should it be "chown --recursive asterisk:asterisk /etc/asterisk" > ? There is reason behind this. It is possibly more secure to make the "owner" root and just allow group access by asterisk. Setting the files as above permits read/write only by the user root and read only by members of the group asterisk. > > 5) Another article says that running as non-root will prevent ToS being > used. > What is ToS? Do I need to be concerned? http://en.wikipedia.org/wiki/Type_of_Service. Why you can't use this as non-root I do not understand... > Any thoughts? > I wrote up my solution for building and running asterisk as non-root here: http://www.theopensourcerer.com/2007/10/30/untangle-asterisk-pbx-and-file-server-all-in-one-part-7/ I have read somewhere that voicemail.conf needs to be writeable by Asterisk so users can change their vmailbox passwords. I haven't confirmed this but I set voicemail.conf to be writeable by group asterisk just in case. Hope this helps. Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie alert: VoIP hardware
Marco wrote: > > Respectfully, I don't agree. I've purchased an "original clone" :-P of > the X100P card, on the long period they almost always have some > drawbacks... Faxing have been troubling for me. Don't know if it was for > the line or else, but with a Digium card I had no problem at all. > No sponsoring in here, ok, but certified hardware works better, > therefore it's a better investment, I think. > I'm just offering my experiences. I have had no problems with my x100p card since using the oslec canceller. There's a big difference between $300 and $34 for one analogue line on a home phone. Of course YMMV ;-) Alan -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie alert: VoIP hardware
Steve Repo wrote: > Hello, > > Please forgive me for i'm not an asterisk user yet. I've done as much > research as I can .. and have the following questions. > > I'm setting up a new office and a home office and i'm shopping for > hardware. > > Office: 2 analog lines > Hardware: TDM412B (2 FXO, 1FXO) > Link: http://www.voipsupply.com/index.php?cPath=99_555_556 > Cost: $303 > > Home: 1 analog line > Hardware: TDM421B (2 FXS, 1 FXO) > Link: http://www.voipsupply.com/product_info.php?products_id=3980 > Cost: $300 If you only have one analogue line why not just get a simple x100p card? When you use OSLEC with them they work great here in the UK. I bought my card from a USA based eBay seller. Total cost for card and shipping was about £17.00 > > Questions: > [1] Can I use oslec for echo cancellation? I'll have beefy hardware. > Is echo cancellation necessary? I would think you will always want to have EC. Whether you will need oslec or not is another matter. If the standard MG2 sounds crap, try oslec. MG2 couldn't deal with echo on my x100p. Oslec is pretty much perfect. I don't think you will need "beefy hardware" either. I have our Asterisk server running on a Via CN700 (1Ghz) along with lots of other applications. No troubles. Of course it is a home box and not heavily used but hey - the mobo only draws about 7W! If you want to know more about it see my sig below. There's a series of articles about setting up and building a home server with Asterisk and other bits and bobs. HTH Alan -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Asterisk as root
Christian wrote: > Hi all, > I have seen discussions on this earlier on, but just want to hear some quick > thoughts. > I am running v1.6 of Asterisk on my Ubuntu installation, I did make config to > make it run at boot. Since I've got a firewall and don't have any other > servers running I am not worried. I have been htinking about running Asterisk > as a seperat user, but haven't done that yet. > Everything is working fine. > What do you think? > Thanks, > Christian > I'd never run a server app as root. It is just asking for trouble IMHO. When I built asterisk on my little custom linux server I documented the process of setting up as a non-privileged process here. Most of the information originally came from the voip-info.org site: http://www.theopensourcerer.com/2007/10/30/untangle-asterisk-pbx-and-file-server-all-in-one-part-7/ Hope this helps. Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote host can't match request NOTIFY???
Grey Man wrote: > On Thu, May 1, 2008 at 7:54 AM, Alan Lord <[EMAIL PROTECTED]> wrote: >> Hi all, >> >> I'm seeing a lot of these messages: >> >> [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: >> Remote host can't match request NOTIFY to call >> '[EMAIL PROTECTED]'. Giving up. >> [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: >> Remote host can't match request NOTIFY to call >> '[EMAIL PROTECTED]'. Giving up. >> > > Asterisk does not correctly match SIP NOTIFY transactions in at least > some cases. Your problem may be related to > http://bugs.digium.com/view.php?id=11848. > > Regards, > > Greyman. > Thanks for that. Not sure I understand it all. I am not actually doing anything when these messages appear. They occur pretty much every minute or so. With or without any calls... Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel Compatibility
Mik Cheez wrote: > Hmph...and it appears no kernel-smp-source exists. You should be able > to compile going to a non-SMP kernel, but there must be a better > solution. I can't believe this hasn't come up before. > > Sorry. You only need the kernel headers in reality I believe. Why not just mail RH and ask them for the headers? Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens Gigaset S685IP Review
Olivier wrote: > Do we agree on the fact you can't change a S68 handset display name (S68 > should be the model name of the handset included in a S685IP package) > from a computer ? > > If my memory serves me right, you can change S685IP base station > settings but not handset settings (display name, subscription to an > other base station, ...), isn't it ? > You can certainly give each handset a more sensible name if that's what you mean... From the web interface to the base station you can do this easily. My three handsets are now called "Alan's", "Helen's" and "Kitchen". You can, according to the manual subscribe each handset to up to 4 different base stations. The actual process of registering the handset requires manually pressing the Page button on the relevant base station, whilst the handset is in "register me" mode... Hope this helps. Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Remote host can't match request NOTIFY???
Hi all, I'm seeing a lot of these messages: [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. They always come in pairs! I've seen a few other posts regarding these - some quite old - but no clear resolution or explanation. Could someone attempt to explain what it means and how I stop it? 10.0.0.2 btw is the IP address of the asterisk server. I'm running Asterisk 1.4.13. Cheers Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens Gigaset S685IP Review
Jaap Winius wrote: However, if you > want to use the unit's integrated answering machine -- in English -- I > would think that would just be a question of finding and installing a > UK firmware version... unless maybe the pre-recorded messages are > separate files. > > Cheers, > > Jaap > I recall reading somewhere that there are "codes" you can enter to change things like this. It might have been on one of the Gigaset forums, but I can't remember. As an aside, I noticed while scanning the back of the manual, that the software, or some if it at least, is LGPL and you can download the source. I've added a link on my blog: http://www.theopensourcerer.com/2008/04/30/siemens-gigaset-uses-lgpl/ Also, for those who struggled with the vcard import, here's a link: http://www.theopensourcerer.com/2008/04/30/s685ip-and-the-vcard-format/ Cheers, Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens Gigaset S685IP Review
Marco wrote: > Hi Alan, > yeah, latest Siemens DECT phones with VoIP support are quite the new > Chuck Norris of cordless phones. Personally I use the C470IP on a > business context and a C475IP at home (for the integrated answering > machine). The audio quality is amazing, and the extra services are > definitely a plus. > I just found 2 big cons: > > * The transfer/hold modes is quite a pain, and it takes many > keypress to activate, which doesn't make them suitable for all people > * The firmware and ALL of the pre-recorded messages are in german. I > had some customers a little scared about this! > > What do you think? > Bye > > Marco Hi Marco, As I've only had them a few days and so haven't used all the features yet. Call transfer is one that I do need to do but, I guess it's the same with most things, once you get into a "routine" I don't think that 2 or 3 keypresses will seem a chore. As for the language, the phones I bought are for the UK market and everything is English. I have heard of Australians, who seem to have to buy from the UK or Europe getting German or Dutch versions before but it wasn't a problem here. These are UK market units. Ciao Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
